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FAQ

What Is VoIP and how does it work?

Voice Over Internet Protocol (also referred to as Internet Telephony) is the transportation of telephone calls over the Internet, connected either via an Internet Adapter that connects your telephone and/or fax to your high speed Internet modem, or by a telephone handset which you purchase that is especially designed to handle IP calls, or by multimedia PCs using PC-to-PC, or PC-to-phone software or even Wi-fi phones. VOIP converts your analogue phone conversation into packets of digital data that are sent across the Internet (just like email or a web page), and then reassembles them into sound at the other end for a crisp, clear telephone experience.

How much bandwidth do I need?
The amount of bandwidth you need is determined by the voice channel and data you use. IP2PBX offers Business DSL, Cable, T-1 or Wireless access to accommodate your traffic requirements.

UNLIMITED Calling Plans - Residential unlimited plans are based on the calling habits of the average individual and is only for use in a home or other residential setting.  IP2PBX retains the right to require account upgrade or suspend service if the calling habits are outside these guidelines.

Business unlimited plans are based on the calling habits of an average small office user sold on a per seat basis.  Business plans are designed for businesses who are not in the business of making and taking phone calls, such as Telemarketers.  Businesses that require multiple users making simultaneous calls or high volume users in the telemarketing business, should contact us and ask about our Bulk Plans.

Analog-to-digital conversion is an electronic process in which a continuously variable analog signal is changed, without altering its essential content, into a multi-level digital signal.

1) An analog signal can be represented as a series of sine waves. The term originated because the modulation of the carrier wave is analogous to the fluctuations of the human voice or other sound that is being transmitted.

2) Analog describes any fluctuating, evolving, or continually changing process.

3) Digital describes electronic technology that generates, stores, and processes data in terms of two states: positive and non-positive. Positive is expressed or represented by the number 1 and non-positive by the number 0. Thus, data transmitted or stored with digital technology is expressed as a string of 0's and 1's. Each of these state digits is referred to as a bit (and a string of bits that a computer can address individually as a group is a byte).

4) Digital technology is primarily used with new physical communications media, such as satellite and fiber optic transmission. A modem is used to convert the digital information in your computer to analog signals for your phone line and to convert analog phone signals to digital information for your computer.

Digital signals propagate more efficiently than analog signals, largely because digital impulses, which are well-defined and orderly, are easier for electronic circuits to distinguish from noise, which is chaotic. This is the chief advantage of digital modes in communications. Computers "talk" and "think" in terms of binary digital data; while a microprocessor can analyze analog data, it must be converted into digital form for the computer to make sense of it.

Compression Algorithms - In computer science and information theory, data compression or source coding is the process of encoding information using fewer bits (or other information-bearing units) than an unencoded representation would use through use of specific encoding schemes. For example, this article could be encoded with fewer bits if one were to accept the convention that the word "compression" be encoded as "comp". One popular instance of compression that many computer users are familiar with is the ZIP file format, which, as well as providing compression, acts as an archiver, storing many files in a single output file.

As is the case with any form of communication, compressed data communication only works when both the sender and receiver of the information understand the encoding scheme. For example, this text makes sense only if the receiver understands that it is intended to be interpreted as characters representing the English language. Similarly, compressed data can only be understood if the decoding method is known by the receiver.

Compression is useful because it helps reduce the consumption of expensive resources, such as disk space or transmission bandwidth.

RTP - The Real-time Transport Protocol defines a standardized packet format for delivering audio and video over the Internet. It was developed by the Audio-Video Transport Working Group of the IETF.

RTP does not have a standard TCP or UDP port on which it communicates. The only standard that it obeys is that UDP communications are done via an even port and the next higher odd port is used for RTP Control Protocol (RTCP) communications. Although there are no standards assigned, RTP is generally configured to use ports 16384-32767. RTP can carry any data with real-time characteristics, such as interactive audio and video. Call setup and tear-down is usually performed by the SIP protocol. The fact that RTP uses a dynamic port range makes it difficult for it to traverse firewalls. In order to get around this problem, it is often necessary to set up a STUN server.

RSVP - A host uses RSVP to request a specific Quality of Service (QoS) from the network, on behalf of an application data stream. RSVP carries the request through the network, visiting each node the network uses to carry the stream. At each node, RSVP attempts to make a resource reservation for the stream.

To make a resource reservation at a node, the RSVP daemon communicates with two local decision modules, admission control and policy control . Admission control determines whether the node has sufficient available resources to supply the requested QoS. Policy control determines whether the user has administrative permission to make the reservation. If either check fails, the RSVP program returns an error notification to the application process that originated the request. If both checks succeed, the RSVP daemon sets parameters in a packet classifier and packet scheduler to obtain the desired QoS. The packet classifier determines the QoS class for each packet and the scheduler orders packet transmission to achieve the promised QoS for each stream.

A primary feature of RSVP is its scalability. RSVP scales to very large multicast groups because it uses receiver-oriented reservation requests that merge as they progress up the multicast tree. The reservation for a single receiver does not need to travel to the source of a multicast tree; rather it travels only until it reaches a reserved branch of the tree. While the RSVP protocol is designed specifically for multicast applications, it may also make unicast reservations.

Quality of Service ( QoS ) refers to control mechanisms that can provide different priority to different users or data flows, or guarantee a certain level of performance to a data flow in accordance with requests from the application program. Quality of Service guarantees are important if the network capacity is limited, especially for real-time streaming multimedia applications, for example voice over IP and IP-TV, since these often require fixed bit rate and may be delay sensitive.

A network or protocol that supports Quality of Service may agree on a traffic contract with the application software and reserve capacity in the network nodes during a session establishment phase. During the session it may monitor the achieved level of performance, for example the data rate and delay, and dynamically control scheduling priorities in the network nodes. It may release the reserved capacity during a tear down phase. A best-effort network does not support Quality of Service.

SIP - the Session Initiation Protocol, is a signaling protocol for Internet conferencing, telephony, presence, events notification and instant messaging.

 


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